Issues 08000 - 08999

[..]
ASTERISK-08000: [branch] Error in CDRs involving bridges and Local/ channels (call files)
ASTERISK-08001: Wrong error message in confiugure script when there is a problem with the openh323 installation
ASTERISK-08002: silence/* files in asterisk-extra-sounds all the same
ASTERISK-08003: [patch] problem with simple transfer of incoming call
ASTERISK-08004: [patch] agi option missing from usage warning
ASTERISK-08005: chan_zap says its ignoring JB settings but it really isn't
ASTERISK-08006: SVN-oej-jitterbuffer-1.2-r38923
ASTERISK-08007: queue.conf option to automatically pause an agent whose status is Busy / In use
ASTERISK-08008: Segmentation Fault when a call is transfered from a queue
ASTERISK-08009: Crash in chan_h323 when dialing invalid non existing extension
ASTERISK-08010: Update to doc/backtrace.txt to reflect 1.4/trunk changes and increase readability
ASTERISK-08011: chan_sip crashes while trying to find video codecs
ASTERISK-08012: SIP Segfault with high call setup volume in ast_rtp_lookup_code()
ASTERISK-08013: SIP paketization
ASTERISK-08014: asterisk 1.2.12.1 crashes with core several times/week during nightly restart script
ASTERISK-08015: asterisk crash when SDP contain no description
ASTERISK-08016: [patch] "list" considered harmful
ASTERISK-08017: [patch] speeling errors in program comments
ASTERISK-08018: Asterisk 1.4 99% cpu usage and crashing
ASTERISK-08019: Segmentation fault on ast_channel_spy_remove
ASTERISK-08020: [patch] rename app_cdr to app_nocdr, update copyright and doxygen info
ASTERISK-08021: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
ASTERISK-08022: [patch] [1.4] Fix "core show translation" output
ASTERISK-08023: RFC broken with media stream on INVITE - Report from Ekiga mailing list
ASTERISK-08024: Disable shell escape '!' (bang) commands in cli
ASTERISK-08025: [patch] Improve Doxygen output, fix various typos
ASTERISK-08026: [patch] ast_gethostbyname("192.168.0.1", &hp) fails to set h_addrtype
ASTERISK-08027: [patch] wav49 format generates 50% compatible files with Windows Media Player
ASTERISK-08028: dot files in /var/lib/asterisk/moh will cause the dir to be treated as a file - blocking the music on hold to play
ASTERISK-08029: [patch] Voicemail recording in progress causes asterisk to freak out when file size exceeds 2.1GB
ASTERISK-08030: macro's appdata can't be longer than 114 chars plus a 3 chars exten
ASTERISK-08031: Zaptel 1.2.10 will not install on Ubuntu 6.10
ASTERISK-08032: [patch] Make redhat init script start asterisk later
ASTERISK-08033: [patch] IAX Peers output over manager should not be CLI formatted
ASTERISK-08034: [patch] CONTROL STREAM FILE AGI command is broken
ASTERISK-08035: [patch] Allow func_curl to substitute channel variables on the passed URL
ASTERISK-08036: Asterisk crashes when logging with cdr_custom
ASTERISK-08037: Queue not advancing to next member
ASTERISK-08038: SIP module deadlocks when PostgreSQL cdr db is overloaded
ASTERISK-08039: last_message_index is returning the first index
ASTERISK-08040: voicemail fails when language set to german (de), 1 message present and no german sound files
ASTERISK-08041: iaxprov.conf in asterisk-1.4.0.beta3 contains tos=lowdelay which is deprecated
ASTERISK-08042: [patch] Added "loose" option to "joinempty" and "leavewhenempty"
ASTERISK-08043: BLF not working when call-limit is enabled
ASTERISK-08044: Client phones nuked with "481 Call leg/transaction does not exist"
ASTERISK-08045: [branch] chan_iax2.c:3782 iax2_trunk_queue: Maximum trunk data space exceeded
ASTERISK-08046: issue 0007268 not properly fixed
ASTERISK-08047: asterisk segfaults at ast_translator_free_path ()
ASTERISK-08048: Dundi Problem with Key generated on hardened system.
ASTERISK-08049: Signalling levels - TBR21 :- the difference level of high frequency group and low frequency shall be 1 ..
ASTERISK-08050: memleak in chan_sip
ASTERISK-08051: After a while of operation, IAX becomes behaving incorrectly, no audio or 1-way, and no-answer
ASTERISK-08052: g729 codec error loading module Asterisk 1.4
ASTERISK-08053: High volume calling results in Asterisk core dump
ASTERISK-08054: The two files produced by the Monitor application have diferent sizes and lengths, which produces unsynchronized recordings
ASTERISK-08055: Voicemail copy to multiple boxes fails
ASTERISK-08056: missing information of the origin exten in the CLI (both with realtime or flat files)
ASTERISK-08057: Caller ID
ASTERISK-08058: Replacing a pointless recursion in q921_transmit()
ASTERISK-08059: [patch] INVITE w/Replaces - Require: header interop issue
ASTERISK-08060: Asterisk not loading properly on boot
ASTERISK-08061: [patch] INVITE w/Replaces - Replaces: header incorrectly uri encoded
ASTERISK-08062: [patch] rtp goes into an error loop on network read error
ASTERISK-08063: [patch] iax channel goes into an error loop on network read error
ASTERISK-08064: SLIN codec noise
ASTERISK-08065: [patch] perl based safe_asterisk
ASTERISK-08066: [patch] callee->context is always empty
ASTERISK-08067: iax2 qualify - false "peer unreachable"
ASTERISK-08068: No such command 'core verbose atleast'
ASTERISK-08069: SendDTMF() through IAX channel transmits only first digit
ASTERISK-08070: URIENCODE doesn't handle '#' correctly. Could be others
ASTERISK-08071: [patch] add checks for calloc calls
ASTERISK-08072: [Patch] unable to specify "no groups" for groups in config files
ASTERISK-08073: Failure getting "online status" on softphone
ASTERISK-08074: [patch] Recording synchronization fails due to bad number of samples correlation in ast_read / ast_write
ASTERISK-08075: Asterisk crash when PRI fails
ASTERISK-08076: Cmd ChanIsAvail() does not return predictable status codes.
ASTERISK-08077: sometimes when user delete all voicemails in their folder, people have trouble getting voicemails
ASTERISK-08078: break instead of continue in get_sip_pvt_byid_locked()
ASTERISK-08079: Asterisk system crashs main server
ASTERISK-08080: app_mixmonitor crashes asterisk
ASTERISK-08081: improper freing of memory in chan_sip.c::unload_module() ?
ASTERISK-08082: ooh323c in asterisk-addons might not be redistributable under the GNU GPL License
ASTERISK-08083: "L" parameter causes asterisk crash
ASTERISK-08084: [patch] Hebrew (he) support in voicemail
ASTERISK-08085: SIP sends no hangup
ASTERISK-08086: PRI de-synchronization
ASTERISK-08087: Asterisk 'invisible' resending registers after registering successful with a peer
ASTERISK-08088: Restore autoframing option
ASTERISK-08089: Frequent Crashes
ASTERISK-08090: "core show version" fails due to comparison of wrong number of args
ASTERISK-08091: CallerID Callwaiting appears to be broken
ASTERISK-08092: [patch] entering a dynamically created conference doesn't observe 'q' (quiet) option
ASTERISK-08093: [patch] nonce-count value is not added correctry
ASTERISK-08094: iax.conf and sip.conf bindaddr don't listen on every interface
ASTERISK-08095: [patch] Refactoring of expression checking implementation
ASTERISK-08096: core show uptime returns only usage information for the comand
ASTERISK-08097: voicemail system sending out mangled emails
ASTERISK-08098: one way audio, when network jitter occur
ASTERISK-08099: wrong menuselect info
ASTERISK-08100: wrong menuselect info
ASTERISK-08101: [patch] A GoSub called from within a Macro clears MACRO_EXTEN, others
ASTERISK-08102: Quality issues with codec g726
ASTERISK-08103: possible memory leak in __sip_ack()
ASTERISK-08104: ExtenSpy segfault on no given argument to spy from.
ASTERISK-08105: asterisk and asterisk-sounds conflicts
ASTERISK-08106: When the phone rings on incoming calls, there are a lot of error messages
ASTERISK-08107: improper handling of sip_pvt references.
ASTERISK-08108: Blind transfers crash Asterisk when verbose = 3
ASTERISK-08109: [patch] progress_setup and progress_alarm are not properly propagated for outgoing H.323 calls.
ASTERISK-08110: T.38 Fallback fails
ASTERISK-08111: call hangup when x=y in L parameter
ASTERISK-08112: Transfers are not bridging properly
ASTERISK-08113: ${IAXPEER(targetchannel)} returns null string when called from a SIP (or other non-IAX2) channel
ASTERISK-08114: Asterisk segfault when trying to include dialplan file with a macro
ASTERISK-08115: [PATCH] chan_iax2.c remove useless #ifdef
ASTERISK-08116: still AC_PROG_LD issues...
ASTERISK-08117: Upon reload of asterisk, cdr_pgsql causes asterisk to seg fault
ASTERISK-08118: MYSQL will allow table LOCK, but error on a UNLOCK
ASTERISK-08119: Outgoing SMS: EMS support broken / UDH-bit not set
ASTERISK-08120: "Dropping voice to exceptionally long"
ASTERISK-08121: Close Codec Translation Feature
ASTERISK-08122: Codec in SDP
ASTERISK-08123: Same that bug 0007351
ASTERISK-08124: [patch][post-1.4] new internal API for CLI
ASTERISK-08125: CLI Command 'module show' is no working
ASTERISK-08126: Shared Line Appearance sla.conf appending - (dash) character on the context? cause segfault
ASTERISK-08127: Call transfer or parking failure
ASTERISK-08128: [patch] H.323 creates channels with nativeformats having MSbit set, which leads to ast_translator_best_choice() chosing it
ASTERISK-08129: asterisk-1.2.13 Postgres support is enabled when it shouldn't
ASTERISK-08130: Segfault when inserting a CDR record with primary key overflowing maximum value
ASTERISK-08131: [patch] Log file rotation on SIGXFSZ doesn't check log file sizes.
ASTERISK-08132: [patch] change app_amd logging of "AMD using the default parameters" from Notice to Debug
ASTERISK-08133: [patch] make init file work in SUSE 10 and Redhat too...
ASTERISK-08134: Reinvite is using local IP of NATed device
ASTERISK-08135: [patch] Asterisk 1.2 Built in transfer works different from version 1.0
ASTERISK-08136: [patch] Warning triggered on "CDR not posted" and "CDR lacks end" when using resetcdr and nocdr apps
ASTERISK-08137: Timelimit functionality is broken in Dial
ASTERISK-08138: [patch] Add accuracy range to incoming distinctive ring match
ASTERISK-08139: Reproduction of bug #6568 in 1.4.0-beta3
ASTERISK-08140: Chanspy application in asterisk 1.4 ver crash the asterisk-segmentation fault (core dumped)
ASTERISK-08141: Asterisk sends CANCEL instead of BYE even if _state is UP
ASTERISK-08142: [patch] Add OSP support
ASTERISK-08143: [patch] Upgrade for atxfer behaviour
ASTERISK-08144: Turning off DTMF Detection or set the sensibility
ASTERISK-08145: Queue Agent joinempty handling missing an option.
ASTERISK-08146: [patch] ast_channel_walk_locked or channel_find_locked can "terminate early"
ASTERISK-08147: Voicemail is leaving open file handles when it create the tmp file
ASTERISK-08148: On call transfer, need to be able to retrieve SIP Referred-by header from the incoming REFER
ASTERISK-08149: Passwords are not saved if voicemail users are in an #include file
ASTERISK-08150: [patch] Remove unused code from manager.c
ASTERISK-08151: music on hold not random
ASTERISK-08152: Forwarding "Moved Temporarily" Not Functioning As Expected
ASTERISK-08153: Potential memory leak in transmit_response_using_temp
ASTERISK-08154: [patch] voicemail playback via odbc connection gives segfault
ASTERISK-08155: wrong behavior of 'L(x:y:z)' parameters in Dial application
ASTERISK-08156: T.38 passthru not invoked when using a Local channel
ASTERISK-08157: can't call new users
ASTERISK-08158: It looks like deadlocking of channel
ASTERISK-08159: powerof(0) can happen with external channel drivers in translate.c
ASTERISK-08160: IAX2 trunking not enabled in one direction for 'user' rather than 'friend'
ASTERISK-08161: Bandwidth Requirement
ASTERISK-08162: inconsistent return checks on handle_request()
ASTERISK-08163: chan_skinny doesn't send keepalives
ASTERISK-08164: potential panics induced by app_dial.c::do_forward()
ASTERISK-08165: "-p: not found" on building
ASTERISK-08166: sh doesn't like the == operator [PATCH]
ASTERISK-08167: call/pickupgroups above 32 do not work, even though the docs state otherwise
ASTERISK-08168: chan_h323 disables fastStart in connect message
ASTERISK-08169: Wrong ptime cause no audio
ASTERISK-08170: Removing "Unavailable Message", "Busy Message", "Name"
ASTERISK-08171: probably useless code in handle_response_register()
ASTERISK-08172: [patch] /proc/zaptel/1 returns spurious characters at end
ASTERISK-08173: Hints no longer work in 1.4beta3
ASTERISK-08174: ParkedCall does not native bridge.
ASTERISK-08175: [patch] app_queue device state change race
ASTERISK-08176: q.931: IntID: Explicit seems not supported by MD110
ASTERISK-08177: [patch] Connect Asterisk as a component to a jabber server
ASTERISK-08178: Limited number of channels
ASTERISK-08179: Limited number of channels
ASTERISK-08180: asterisk crashes when transfering zap channel from idefisk to park extension 700
ASTERISK-08181: [PATCH] if (debug); printk ... in wcte11xp.c
ASTERISK-08182: Fix make when make is not GNU make
ASTERISK-08183: Asterisk process utilizing 100% of CPU time after a P2P'd sip call is hung up
ASTERISK-08184: Background() application over AGI doesn't return control until file is through playing
ASTERISK-08185: Called SIP subscriber still ringing even after hanging up the call by calling side
ASTERISK-08186: Using ODBC voicemail, user intros for unavail, busy and greet are not entered in database
ASTERISK-08187: IMAP storage in trunk truncates username and password to 3 chars
ASTERISK-08188: When using IMAP storage, "voicemail show users" does not read message count from imap
ASTERISK-08189: Potential deadlock in zt_hangup() (with driver interaction...)
ASTERISK-08190: [patch] OriginateSuccess and OriginateError incomplete
ASTERISK-08191: sip show inuse does not return anything
ASTERISK-08192: reference memory after free(). in pbx/pbx_spool.c
ASTERISK-08193: unable to add new trunks in asteriskNOW
ASTERISK-08194: bkps directory missing -- tarball creation fails
ASTERISK-08195: install/runtime support for 586 and lower
ASTERISK-08196: memory leak , ast_frdup called without free and passthrough codecs used.
ASTERISK-08197: linux/compiler.h must not be included in 2.6.17
ASTERISK-08198: bug with INVAL event
ASTERISK-08199: Asterisk is sending INVAL packages without a reason
ASTERISK-08200: [patch] forkcdr does not work as expected
ASTERISK-08201: 1.4.0b3 crashed during call transfer
ASTERISK-08202: Transfer on a Polycom phone does not set hint to Idle on transfer completion
ASTERISK-08203: /dev/zap/pseudo permissions can't be changed
ASTERISK-08204: Voicemail password problem with users.conf
ASTERISK-08205: Chanspy whisper does not work as expected
ASTERISK-08206: Top-level Makefile variable not exported
ASTERISK-08207: Agents and SIP attended transfers gives warning messages (codec issue)
ASTERISK-08208: Escaped SIP URI (RFC 3261) doesn't match in dialplan (e.g. "#" with SNOM phones)
ASTERISK-08209: [patch] Forwarding old voicmail to another user will just be sent to email not to target voicemail
ASTERISK-08210: G722 audio tarball filename failure for "make"
ASTERISK-08211: SRV record lookup failing
ASTERISK-08212: The channel is not hanged up in the right time
ASTERISK-08213: unexpected ringtone
ASTERISK-08214: [patch] Allow whisper functionality in Meetme
ASTERISK-08215: No option in GUI to modify codec prefs in iax.cong and sip.conf
ASTERISK-08216: [patch] Packet2Packet bridge incompatible with STUN packets processing
ASTERISK-08217: memory leak at res_features.c
ASTERISK-08218: Update sip.conf.sample files to show usage of port, and different between bindport
ASTERISK-08219: [patch] clean up some compile issues on FreeBSD (6.1)
ASTERISK-08220: Admin password not set for VMplayer version
ASTERISK-08221: Digium URLs in the admin UI are broken
ASTERISK-08222: Dialog box for mismatched admin passwords refers to "root" password
ASTERISK-08223: Localhost login refers to "AsteriskNow" not "AsteriskNOW"
ASTERISK-08224: Scrollbox on Time Setup screen only one row high; scrollbar unusable
ASTERISK-08225: AsteriskNOW logo splash screen needed on boot screen
ASTERISK-08226: [patch] Zaptel trunk fail to compile / install on FreeBSD (6.1)
ASTERISK-08227: No root password (security problem)
ASTERISK-08228: Input field validation is timing dependent
ASTERISK-08229: Input fields consisting of all spaces are allowed
ASTERISK-08230: Overloaded use of user "password"
ASTERISK-08231: System allows variable length extensions
ASTERISK-08232: Sort-by-name option in user list
ASTERISK-08233: Type-ahead search in user list
ASTERISK-08234: Tooltip for Conference , "Record conference" is missing
ASTERISK-08235: Populate call queue extension with next available extension number
ASTERISK-08236: Make all users agents by default
ASTERISK-08237: Graceful handling of case where no analog lines are installed
ASTERISK-08238: On Trunks: "creating new entry" under wrong list
ASTERISK-08239: Trunks: empty VoIP provider list
ASTERISK-08240: For SIP providers, help choose closest server
ASTERISK-08241: System information
ASTERISK-08242: System information: need GUI version information
ASTERISK-08243: Backup can create multiple backup files with the same name
ASTERISK-08244: Backup files can't be downloaded / uploaded
ASTERISK-08245: Backup: "download configuration backup"
ASTERISK-08246: Optionstab: typos "atleast" and "donot"
ASTERISK-08247: [bounty] feature request - QSIG call diversion interop with SIP
ASTERISK-08248: Support for uploaded files
ASTERISK-08249: System info tab should provide access to logs of recent error/status messages
ASTERISK-08250: Clicking on a backup filename reports "404 file not found"
ASTERISK-08251: No way to restore a backup once created
ASTERISK-08252: Backup files should summarize the date the backup was created
ASTERISK-08253: sysinfo displays incorrect information
ASTERISK-08254: AgentCallbackLogin and SIP hold music doesn't work in user to agent direction.
ASTERISK-08255: [patch] fixed a typo in the 'dundi-e164-canonical' section
ASTERISK-08256: Custom-voip provider has provider = iaxtel
ASTERISK-08257: [patch] compiling and installing asterisk-gui on FreeBSD (6.1)
ASTERISK-08258: No DTMF tone with chan_misdn
ASTERISK-08259: [patch] asterisk-addons doesn't respect --prefix
ASTERISK-08260: having a video play for auto attendant for those with video phones
ASTERISK-08261: unable to log in to gui using konqueror web browser
ASTERISK-08262: [patch] Bringing back to Makefile pridump, pritest and testprilib binaries
ASTERISK-08263: panels do not respond to mouse click in konqueror
ASTERISK-08264: [patch] snmp.txt says it needs more libraries, but doesn't tell us which ones
ASTERISK-08265: [patch] odbcstorage.txt references command in voicemail.conf that is not in .sample file
ASTERISK-08266: asterisk-gui setup.html crash asterisk with included file on extensions.conf from nfs
ASTERISK-08267: chan_gtalk and chan_jingle outgoing calls remain locked without calling anything
ASTERISK-08268: asterisk-ooh323c from asterisk-addons svn trunk don't compile with latest asterisk svn trunk
ASTERISK-08269: Crash when having more than 10 IAX registrations per second
ASTERISK-08270: [patch] ES-EN translation of code comments in fskmodem.c
ASTERISK-08271: On high call volume, Asterisk starts reporting: cause 34 - Circuit/channel congestion
ASTERISK-08272: [patch] fix check for curl-config and removed reduntant code
ASTERISK-08273: Webserver is saying "Invalid/Unknown Command" instead of saying "already Logged In"
ASTERISK-08274: Use of voicemail ODBC storage not possible with Postgresql
ASTERISK-08275: Possible issue with Early Media still in invitestate-1.4 branch (and thus other Asterisk branches)
ASTERISK-08276: [patch] Voicemail does not playback via ODBC voicemail storage with Postgresql database
ASTERISK-08277: /etc/init.d/asterisk and/or safe_asterisk breaks cron/anacron (Debian)
ASTERISK-08278: [patch] voicemail with volgain leaves behind temp file
ASTERISK-08279: [patch] add option to disable announcement of queue position
ASTERISK-08280: [patch] aditional manager commands DBDel and DBDelTree
ASTERISK-08281: Enhance CUT function to allow range variable
ASTERISK-08282: Goto fails in applicationmap (res_features)
ASTERISK-08283: Asterisk crashes when updating state on a expired realtime peer (res_config_mysql)
ASTERISK-08284: app_dial setting callerid to ${EXTEN} too soon
ASTERISK-08285: Reading DTMF fails on IAX2
ASTERISK-08286: asterisk-gui v150 fails to "make" under OpenSUSE 10.1 64-bit
ASTERISK-08287: extensions limited to 4 digits in "updated" AsteriskNOW
ASTERISK-08288: [patch] Via: header may contain multiple values
ASTERISK-08289: Permission denied, opening /dev/snd/controlC0, even if the user, under which asterisk runs as, has access to it.
ASTERISK-08290: [patch] Incorrect variable name 'rtignoreexpire' in iax.conf.sample
ASTERISK-08291: multiples refers problem
ASTERISK-08292: 487 retransmits are not ACKed
ASTERISK-08293: Inbound Caller Does Not Hear Ringing
ASTERISK-08294: [patch] Say Digits does not work correctly
ASTERISK-08295: STRFTIME() requires an argument, but function description does not reflect that
ASTERISK-08296: [patch] extend IAX2 to support OSP protocol
ASTERISK-08297: Manager redirect hangs up on calls in AGI
ASTERISK-08298: Asterisk server crashes on 'undefined symbol: ast_adsi_available' in VoiceMailMain if res_adsi.so is not loaded
ASTERISK-08299: Need FreeWorld Dialup, Broadvoice, Sipura install procedures for AsteriskNOW
ASTERISK-08300: T.38 negotiation fails when h263 is enabled
ASTERISK-08301: [patch] Caller ID not set in CDR
ASTERISK-08302: zaptel module (wcte11xp) makes asterisk unable to reproduce audio messages
ASTERISK-08303: Hard phone issue:
ASTERISK-08304: [patch] segmentation fault when unable to play files
ASTERISK-08305: [patch] When using L option on Dial, instead of warning asterisk disconnects the call
ASTERISK-08306: Incoming RDNIS redirecting number variable left unset on EuroISDN
ASTERISK-08307: [patch] OpenBSD 4.0 "make" fails
ASTERISK-08308: Hard phone issue:
ASTERISK-08309: Asterisk Ignoring port parameter in sip.conf
ASTERISK-08310: [patch] DTMF fails and one-way-audio after negative timestamp.
ASTERISK-08311: Asterisk dumps core when briding in p2p mode
ASTERISK-08312: Playback and StopPlayback Manager commands
ASTERISK-08313: Queue does not work with SIP gateway
ASTERISK-08314: Need IE, Safari support
ASTERISK-08315: [patch] ast_app_getdata() without any prompts to play
ASTERISK-08316: manager: split / redirect call
ASTERISK-08317: show iax peer details in manager and cli
ASTERISK-08318: [patch] introduce distinction between overlap-dial in sending/receiving mode
ASTERISK-08319: even though configure is ran with --prefix, make install tries to mkdir /var/lib/asterisk
ASTERISK-08320: Large SIP messages are truncated to 4096 bytes
ASTERISK-08321: 'admin' console login password not set in vmplayer version
ASTERISK-08322: "Local extension settings" should be on the Admin options panel
ASTERISK-08323: Download backup should be streamlined
ASTERISK-08324: "next available" extension numbers should conform to extension length
ASTERISK-08325: Misspelling in Service providers tab
ASTERISK-08326: Asterisk stays in the audio path if "t" option in Dial is used
ASTERISK-08327: Action originate with app Set don't call
ASTERISK-08328: Asterisk crashes if it can not find file that it just recorded
ASTERISK-08329: Authentication using contact user from registration instead of specified
ASTERISK-08330: Adding Rhino Card Installation to Core Installer
ASTERISK-08331: Email notification has blank CIDNAME (should use CIDNUM or an "unknown caller" if empty but doesn't).
ASTERISK-08332: [patch] allow asterisk database to be used for static configuration for files
ASTERISK-08333: [patch] implement basic "Shared Lines" functionality
ASTERISK-08334: Chanell always showsa as Zap/0-0 regardless of actual channel in use.
ASTERISK-08335: REFER not working with Cisco hardware when doing local attended call transfer
ASTERISK-08336: SIP HOLD propegation to AST_CONTROL
ASTERISK-08337: Asterisk core when busy in a zap call
ASTERISK-08338: Call processing stops during reload on systems with large dialplan
ASTERISK-08339: [patch] MWI Error with NOTIFY on Cisco IP phone firmware > 8.0.3
ASTERISK-08340: Spandsp + Ast 1.4B3 - crash on incoming RXFAX
ASTERISK-08341: Unable to join queue
ASTERISK-08342: Asterisk randomly crashes
ASTERISK-08343: Reg asterisk-1.4 installation problem-undefined rerference to'ast_copy+_string'
ASTERISK-08344: [patch] Limit on simultaneous calls for queue members
ASTERISK-08345: building 1.4.0-beta4 fails when linking chan_zap (missing libpri dependency alert)
ASTERISK-08346: [patch] Update to use the new ast_channel_alloc format
ASTERISK-08347: Local channels hanging
ASTERISK-08348: Call waiting Notification from PRI
ASTERISK-08349: [patch] extend app_SMS to support protocol 2 (in use in Italy, Spain, xxx)
ASTERISK-08350: [patch] Caller Id and Message Waiting Indicator problems
ASTERISK-08351: Display error in Agents list
ASTERISK-08352: Goto Exten fails when used as a step in Voice Menus
ASTERISK-08353: Festival Application Hangs Call
ASTERISK-08354: asterisk drops call long call
ASTERISK-08355: chan_sip does not handle 504 "Service Unavailable" case
ASTERISK-08356: [patch] dstchannel in cdr is empty when transfer call
ASTERISK-08357: [patch] make field names configurable in cdr_addon_mysql
ASTERISK-08358: [patch] Wrong Coding of Name in 'To:' header of Emails
ASTERISK-08359: [patch] Asterisk doesn't send CANCEL before Ringing
ASTERISK-08360: SIP, dtmf-relay, feature key presses being ignored
ASTERISK-08361: RetryDial does not properly support the G() Dial option
ASTERISK-08362: SIP bug in handling invitestate
ASTERISK-08363: [branch] improper computation of Content-Length in add_t38_sdp()
ASTERISK-08364: [patch] logic of handle_common_options() in channel_sip.c (2 issues)
ASTERISK-08365: [patch] C++ modules fails to compile, strings.h is not C++ clean
ASTERISK-08366: Meetme conference application randomly crashing with app_meetme.c errors
ASTERISK-08367: [patch] threads syncronization
ASTERISK-08368: ./configure --prefix... ignored.
ASTERISK-08369: No OriginateSuccess or OriginateFailure event after a Originate command.
ASTERISK-08370: Add link in GUI directly to bug tracking system to report problems
ASTERISK-08371: Support 12-hour format clock (in addition to 24-hour/military clock currently supported)
ASTERISK-08372: Allow for GUI updating from within the GUI
ASTERISK-08373: Sound for "October" is said as "Tober"
ASTERISK-08374: [patch] likely memory leak in app_dial (trunk, 1.4 and 1.2)
ASTERISK-08375: [patch] Added RTCP Manager Events to rtp.c
ASTERISK-08376: Asterisk 1.4.0-beta4 compile errors on Fedora Core 6
ASTERISK-08377: "System Info" tab reports "file not found" dialog box
ASTERISK-08378: System Info / Logs report "404 Not Found"
ASTERISK-08379: Typo on screen 1 of 7 of setup wizard "Verify Analog ports"
ASTERISK-08380: "starting point" of allocation should be consistent with extension length selection
ASTERISK-08381: Setup wizard: Calling rules step (5 of 7); should default to provider
ASTERISK-08382: Setup calling rules (5 of 7): Can't "save" edits to calling rules
ASTERISK-08383: Tab style in "Options" and "System Info" should match
ASTERISK-08384: Calling Rules -- dialog box cascade for "undefined" items
ASTERISK-08385: "F2 Incorporated"'s logo is broken
ASTERISK-08386: [patch] make message length configurable per user instead of only globally
ASTERISK-08387: [patch] Allow voicemail to use an external app and smdi at the same time.
ASTERISK-08388: mis-spelling in doc/snmp.txt
ASTERISK-08389: build fails on snmp/agent.c
ASTERISK-08390: RFC 2833 dialtone packets out of order can cause extra digits to be reported
ASTERISK-08391: Dialog box for missing password on login has wrong text
ASTERISK-08392: Several problems with setup wizard login screen
ASTERISK-08393: Possible to drive UI to state strange state
ASTERISK-08394: Copyrights need updating
ASTERISK-08395: Updated text needed for "Service Providers" tool tip
ASTERISK-08396: Incoming calls tab is blank
ASTERISK-08397: Admin password screen still refers to "Business Edition"T
ASTERISK-08398: System failed to detect analog ports
ASTERISK-08399: asterisk fails to cross compile for arm
ASTERISK-08400: Bug 0006181 still exists.
ASTERISK-08401: iax2 crash on transfer
ASTERISK-08402: Trademark link inside the footer is broken
ASTERISK-08403: setup wizard stuck on step 2
ASTERISK-08404: Legal Information link broken
ASTERISK-08405: Problem receiving calls from BroadWorks
ASTERISK-08406: IAX2 outgoing calls not working
ASTERISK-08407: Music on Hold for Call Queues
ASTERISK-08408: Asterisk segfaults (core dumped) at startup because of corrupted astdb
ASTERISK-08409: The UI should report to the user attempted use of incompatible browsers
ASTERISK-08410: Active channels not cleaned upon entering an UNREACHABLE state
ASTERISK-08411: No inbound CallerID when Distinctive Ring Detection is enabled.
ASTERISK-08412: No dialtone on analog FXS ports
ASTERISK-08413: Dead AGI : not able to use StartMusicOnHold application
ASTERISK-08414: Can't make rpm in Asterisk 1.2.14 due to missing files in asterisk.spec
ASTERISK-08415: Remove silence files from Asterisk Extra Sounds, since Core sounds have them since 1.4.b4
ASTERISK-08416: Pickup using g729
ASTERISK-08417: [patch] canreinvite = nonat does not cause packet2packet bridge
ASTERISK-08418: [patch] libiax2 wrong timestamp
ASTERISK-08419: Parking causing crashes
ASTERISK-08420: segfault at irregular interval
ASTERISK-08421: [patch] Add a jabber text receiver application, and make Asterisk a Gtalk to phone gateway
ASTERISK-08422: Possible memory leak doing only inbound SIP handling
ASTERISK-08423: Cut function requires | delimiter
ASTERISK-08424: Parked Calls drop immediately
ASTERISK-08425: [patch] enable MPEG4 Part 2 video codec pass-through
ASTERISK-08426: [patch] option to restrict manager users to a single simultaneous login
ASTERISK-08427: func_math.c: iaction set to GTFUNCTION when the first char of operator is '='
ASTERISK-08428: TDM400+SIP paketizations
ASTERISK-08429: Stringfield pool corruption, segmentation fault during free.
ASTERISK-08430: Blind Transfer does not fail when destination is unreachable
ASTERISK-08431: Asterisk auto-creates meetme conference when not ask to do so.
ASTERISK-08432: [patch] libpri Makefile doesn't use full path to restorecon
ASTERISK-08433: [patch] zaptel Makefile doesn't use full path to restorecon
ASTERISK-08434: MOH doesn't resume from where it was left off
ASTERISK-08435: [patch] app_page.so 's' option flag - skip adding channel to meetme if devicestatus != not in use
ASTERISK-08436: vm-youhaveno sound is missing in spanish
ASTERISK-08437: Sound 1M is not found in Spanish even though it exists
ASTERISK-08438: [patch] Shell Dialplan Function, returns output
ASTERISK-08439: Asterisk does not reinvite peer for G.711 after T.38 negotiated failed with a "488" Event
ASTERISK-08440: GetConfig + #include causes segfault
ASTERISK-08441: [patch] coding guidelines compliance for main/*.c
ASTERISK-08442: billsec is 0 even when the call is answered
ASTERISK-08443: Asterisk Manager Interface Reload segfaults if clients are connected.
ASTERISK-08444: 1.4.0 release UPGRADE.txt lists old show channels concise method whcich doesn't work
ASTERISK-08445: Blind Transfer within DIAL in DeadAGI does not work
ASTERISK-08446: [patch] Fix bad handling of #include directives from manager GetConfig/UpdateConfig
ASTERISK-08447: * eats all CPU.
ASTERISK-08448: Crash on blind transfer of an incoming call (queue)
ASTERISK-08449: Asterisk 1.2.7.1 Crashing
ASTERISK-08450: Check type selection for sizes
ASTERISK-08451: Frequent seg fault in ast_cdr_alloc() at cdr.c:438
ASTERISK-08452: verbose info with no if (option_debug) so it always shows
ASTERISK-08453: /etc/sudoers file becomes corrupt
ASTERISK-08454: Zap lines are always named as Zap/0-0
ASTERISK-08455: Lithuanian syntax for ast_say_number_full
ASTERISK-08456: [patch] SQLite3 CDR Backend
ASTERISK-08457: IAX2 appears to deadlock on OS X
ASTERISK-08458: Cannot pass audio after transfer, intermittent segfault
ASTERISK-08459: pbx_load_users adds IAX instead of IAX2 to dial string and hint
ASTERISK-08460: Segfault on 1.4.0 with E1s
ASTERISK-08461: [patch] openh323 of Debian not detected by autoconf
ASTERISK-08462: SIP User-Agent string should display Asterisk version
ASTERISK-08463: IAX2 configuration parser reverses general and specific parameters when loading users
ASTERISK-08464: Sounds played in MEETME_EXIT_CONTEXT of conference stutters.
ASTERISK-08465: app_wait rounds down delay to nearest whole number
ASTERISK-08466: queue causes a crash
ASTERISK-08467: Call dropped with "FRAME_CONTROL (5)" message
ASTERISK-08468: Crash 20 seconds after reload app_queue.so
ASTERISK-08469: Replace not working properly?
ASTERISK-08470: 302 Redirect not working?
ASTERISK-08471: Extensions. Only digits allowed in GUI, other characters ok if file users.conf.
ASTERISK-08472: random clicking of pages in GUI produces a crash in manager
ASTERISK-08473: Asterisk GUI does not output Asterisk Log
ASTERISK-08474: AsteriskGUI tells wrong version in system info tab
ASTERISK-08475: Select MoH thru GUI
ASTERISK-08476: Calling Rules Limited to Dialpan1
ASTERISK-08477: Transfering of calls does not work in 1.4 through chan_agent
ASTERISK-08478: MusicOnHold application drops call after exactly one minute
ASTERISK-08479: mixminotor don't record sounds played to callee
ASTERISK-08480: Seg fault on call made after manager connection
ASTERISK-08481: Asterisk 1.4 crash for some reason (iax stuff)
ASTERISK-08482: Attend transfer with internal Polycom tranfer method does not show original caller id
ASTERISK-08483: Origdate field in voicemail msg0000.txt has an extra space
ASTERISK-08484: Asterisk crashes in bridge_p2p_rtp_write
ASTERISK-08485: [patch] calltime duration with L() keeps warning for all last seconds
ASTERISK-08486: file include directive doesn't work in extensions.conf
ASTERISK-08487: Dynamically Generate list of sound files
ASTERISK-08488: codec_zap no longer compiles as it can't find zaptel/zaptel.h in nonstandard place
ASTERISK-08489: Users can't make IAX2 calls
ASTERISK-08490: wcte11xp and wcte4xxp and wctdm inside in one box work issue
ASTERISK-08491: Stack gets confused when Protocol error received after SETUP
ASTERISK-08492: Question about the Issue 5853
ASTERISK-08493: Can't compile with debian unstable due to genksyms not found
ASTERISK-08494: [patch] RELEASE COMPLETE is sent in state ACTIVE - leaves the trunk busy
ASTERISK-08495: Assign extensions to queues
ASTERISK-08496: [patch] transmit_state_notify for DIALOG_INFO_XML wrong
ASTERISK-08497: asterisk reinvites to G.711 after a T.38 negotiation - fax fails depending on ATA config
ASTERISK-08498: Allow a reason to be specified when pausing an agent.
ASTERISK-08499: [patch] Allow a reason to be specified when pausing an agent.
ASTERISK-08500: [patch] zapata.conf cannot reset usedistinctiveringdetection and distinctiveringaftercid
ASTERISK-08501: [patch] hasmanager in users.conf has no effect
ASTERISK-08502: [branch] PlayDTMF with a non-existing channel will cause segmentation fault
ASTERISK-08503: Console interface (-r) stops operating, with logging
ASTERISK-08504: When using the asterisk manager to originate a call the billsec field in CDR's is set to 0
ASTERISK-08505: 1.4.0 has teardown/hangup issues after attended transfer
ASTERISK-08506: [patch] Fix app_read to play multiple files
ASTERISK-08507: Registration acknowledgement incorrectly handles missing refresh value
ASTERISK-08508: Asterisk 1.4.0 Hint status not updated afer HOLD state
ASTERISK-08509: callerid.c loses name when returning PRIVATE_NUMBER flag
ASTERISK-08510: Inactivity TimeOut => panel opens blank frame
ASTERISK-08511: Sip phones don't appear in users list
ASTERISK-08512: Attended Transfer with Polycom handset results in stuck call.
ASTERISK-08513: Compile of SVN zaptel failed
ASTERISK-08514: Codec_zap SVN failed to build
ASTERISK-08515: ALSA output causes Seg Fault
ASTERISK-08516: ztcodec not in menuselect selection
ASTERISK-08517: Asterisk crashed with core dump in reqprep
ASTERISK-08518: Spying by an IAX2 endpoint kills SIP calls
ASTERISK-08519: password changes in voicemail does not work (using realtime)
ASTERISK-08520: "Zaptel transcoder support loaded" message messes up screen and obscures button
ASTERISK-08521: Wav49 support appears broken
ASTERISK-08522: [patch] Voicemail fails to authenticate users created from users.conf
ASTERISK-08523: libnsl no necesary in BSD's
ASTERISK-08524: ZAP channels dies after a while (a week or close) of asterisk usage
ASTERISK-08525: setMusiconHold option does not show MOH files.
ASTERISK-08526: Asterisk crashes in call centre environment several times a week
ASTERISK-08527: When configured with hasiax = yes IAX is dialed instead of IAX2
ASTERISK-08528: asterisk will not handoff RTP!!!
ASTERISK-08529: SIP message 420 Bad extension sent out malformed.
ASTERISK-08530: MeetMeJoin manager event sent when leaving room
ASTERISK-08531: Extension length problems with voice mail
ASTERISK-08532: System happy to assign multiple extensions to the same analog line
ASTERISK-08533: "Called id" text in user extensions setup wizard should be "caller id"
ASTERISK-08534: Asterisk fails to build from 1.4 branch on Fedora Core 5 due to a missing gsm.h
ASTERISK-08535: Attended transfers to parking broken
ASTERISK-08536: asterisk config options for external libs (ssl, qt3, ncurses, ...) are not followed
ASTERISK-08537: Latest SVN crashes asterisk with SIP registion and res_config_pgsql.(pgsql)
ASTERISK-08538: Module cdr_manager.so doesn't load if cdr_manager.conf exists.
ASTERISK-08539: brazilian portuguese syntax in voicemail with problems (pt_BR)
ASTERISK-08540: Crash with queues and transfers on a call center eviroment.
ASTERISK-08541: Cannot create incoming call route in AsteriskNOW - GUI does not store it.
ASTERISK-08542: G729 No path to translate.
ASTERISK-08543: G729 No path to translate.
ASTERISK-08544: Code error in chan_sip.c
ASTERISK-08545: Calls coming on off of a AgentCallbackLogin() queue get dropped upon transfer
ASTERISK-08546: GUI keeps kicking users out after getting to wizard
ASTERISK-08547: [patch] Properly handle tempates on config read/write
ASTERISK-08548: Audio files incorrect for SayNumber with 13..14..15..
ASTERISK-08549: Attended transfers using Polycom phones create 100% CPU utilization
ASTERISK-08550: [patch] SQLite3 resource
ASTERISK-08551: Segmentation fault (core dumped) when try to use the g729 codec
ASTERISK-08552: PIN Code and Administrator PIN Code Conferencing are ignored for app_meetme
ASTERISK-08553: SVN rev 1806 fail on compile
ASTERISK-08554: SIP message 420 Bad extension sent out malformed.
ASTERISK-08555: Channel h323 has error in code (invalid function using -> ast_append_ha)
ASTERISK-08556: Incorrect parsing of IAX2 video frames
ASTERISK-08557: [patch] performing a goto / gotoif / gotoiftime in the h extension changes the dst field of the cdr
ASTERISK-08558: chan_sip doesn't recognize localnet
ASTERISK-08559: [patch] While dtmfmode set to inband asterisk still negotiates rfc2833 mode, but fail to recognize digits
ASTERISK-08560: hints on SIP-accounts don't work
ASTERISK-08561: Check PEER first
ASTERISK-08562: Iaxy wont give dial tone
ASTERISK-08563: [patch] ast_load_realtime calls ast_load_realtime_all with wrong parameters
ASTERISK-08564: DTMF no longer recongnized on ParkedCalls
ASTERISK-08565: Manager Interface no longer recognizes the Login action
ASTERISK-08566: Last few daily builds of 1.4-svn have produced distorted audio in SIP
ASTERISK-08567: DTMF outpulse bug
ASTERISK-08568: [patch] two minor bugfixes for voicemail (one for IMAP storage, one for email notifications)
ASTERISK-08569: IMAP storage does not work with c-client 2006
ASTERISK-08570: unload res_snmp will cause a segfault
ASTERISK-08571: Chan H323 will not load
ASTERISK-08572: Channel ooh323 will not appear in channeltypes
ASTERISK-08573: [patch] Store vm password in external file
ASTERISK-08574: including a shipped header file based on includepath search doesn't make sense
ASTERISK-08575: dtmf digits are not transmited reliably from h323
ASTERISK-08576: Outbound DTMF Fails with Sipgate using RFC2833
ASTERISK-08577: No more registry after a timeout
ASTERISK-08578: Dial() sending to Macro upon connect
ASTERISK-08579: [patch] imap storage does not work in conjunction with realtime voicemail
ASTERISK-08580: During the install process, "make samples" assumes that you have installed the gsm sound files
ASTERISK-08581: Asterisk doesn't use externip if private IP is outside localnet
ASTERISK-08582: [patch] GetGroupCount causes a Seg Fault!
ASTERISK-08583: GUI Voicemenu does not retain keypress 'GoToMenu' events (Can't get it to hold on to a menu tree.)
ASTERISK-08584: [patch] Remote (called) Party Identification - chan_sip & chan_skinny implementation
ASTERISK-08585: Queue commands 'i' option
ASTERISK-08586: Wizard wiped out call rules on first setup
ASTERISK-08587: [patch] allow the * as the exit dtmf
ASTERISK-08588: [patch] Know on which channel a stream occurs
ASTERISK-08589: Add g729 pass-thru support for Sigma Designs boards support
ASTERISK-08590: accountcode variable is not passed forward on a transfer
ASTERISK-08591: RFC2833 DTMF "Zero" missing inner RTP packets
ASTERISK-08592: Additional checks for option_debug
ASTERISK-08593: r50032 broke DTMF (rfc2833)
ASTERISK-08594: 500 ms delay on answer introduced channel locking issue
ASTERISK-08595: Race condition in app_meetme when using the 'e' (empty) and 'd' (dynamic) option and two calls arrive at the "same" time.
ASTERISK-08596: ael2 reload nukes subscription tables
ASTERISK-08597: Addition of Transfer and Flash-Hook-Transfer functions into Gui
ASTERISK-08598: Voice Menu Config does not retain extensions
ASTERISK-08599: mohsuggest doesn't work
ASTERISK-08600: ael2 does not support extension in macro
ASTERISK-08601: Seg.fault when parking a call via an extenension
ASTERISK-08602: not able to send/receive calls after upgrade
ASTERISK-08603: [patch] In Realtime Queues, dynamic queue members do not always load the members correctly
ASTERISK-08604: d-channel is hardcoded to 24 and 16 for PRI T1 and E1 respectively
ASTERISK-08605: tab isnt working for module load
ASTERISK-08606: [patch] core show channeltype foo
ASTERISK-08607: Using Mixmonitor app eat's a lot of memory on > = 10 Simultaneous call
ASTERISK-08608: [branch] First patch for the CLI filtering
ASTERISK-08609: Core dumped when too many calls are queued
ASTERISK-08610: memory leak on command line completion
ASTERISK-08611: manager authentication does not work
ASTERISK-08612: many service provider - missing scrollbar
ASTERISK-08613: contact information should be in users.conf - no incoming calls
ASTERISK-08614: nat=no is not RFC 3261 compliant regarding sending responses
ASTERISK-08615: Remote-Party-ID
ASTERISK-08616: [branch] Don't rotate ODBC log
ASTERISK-08617: Voicemail recordings not getting saved
ASTERISK-08618: [patch] Add option to revert old ChanIsAvail() with 's' option behavior
ASTERISK-08619: GUI - "Buy Now" button non-functional
ASTERISK-08620: GUI - "Incoming Call Rule" Notice stays after configuration
ASTERISK-08621: GUI - "Calling Rules" Pattern match adds an extra digit
ASTERISK-08622: GUI - "Calling Rules" save option does not appear if you only edit the "follow by [ ] digits" box
ASTERISK-08623: [patch] new channel driver : chan_unistim for Nortel Unistim IP phones
ASTERISK-08624: Dell 2950 - AsteriskNow will not install using gui
ASTERISK-08625: Voicemail w/ odbc fails under certain circumstances
ASTERISK-08626: List of prompts with cut off audio
ASTERISK-08627: Installer Static IP check 'fails' valid IP.
ASTERISK-08628: iwconfig missing from install.
ASTERISK-08629: Wrong 'TYPE' used for wireless card in ifcfg-ethx
ASTERISK-08630: Voicemail with IMAP storage sends two email messages
ASTERISK-08631: "BuyNow" button is not fixed on the screen
ASTERISK-08632: "BuyNow" button does not appear to link to a working Web page
ASTERISK-08633: Service provider password field shows password as clear text
ASTERISK-08634: Outbound calling doesn't work on GUI / Asterisk NOW
ASTERISK-08635: System info / log doesn't filter out extraneous characters that may affect HTML formatting
ASTERISK-08636: Incoming call rules table not correctly formatted immediately after adding a rule
ASTERISK-08637: Incoming call rules table not correctly formatted immediately after adding a rule
ASTERISK-08638: Incoming call rules table not correctly formatted immediately after adding a rule
ASTERISK-08639: Incoming call rule to route to an extension does not appear to work
ASTERISK-08640: Accessing HTML Manager Event Interface causes Segmentation Fault
ASTERISK-08641: TYPO: Makefile looks for .bz instead of .gz
ASTERISK-08642: Accessing HTML Manager Event Interface causes Segmentation Fault
ASTERISK-08643: Recording mixing problem with asterisk
ASTERISK-08644: Calling Rules no longer display since deleting 1 rule - Asterisk Now GUI
ASTERISK-08645: Asterisk sometimes hangs up channel on Manager Redirect command
ASTERISK-08646: Asterisk crashes randomly when a transfers from queue member to an SIP extension with spies attached.
ASTERISK-08647: Zaptel tone indication for congestion wrong for tonezone .au
ASTERISK-08648: Segfault at CLI
ASTERISK-08649: [patch] Variable used after it was free()
ASTERISK-08650: * segfault on module unload chan_sip.so
ASTERISK-08651: stray DTMF (rfc2833) during conversations
ASTERISK-08652: doens't start moh when on hold
ASTERISK-08653: [patch] remove 2 identical checks.
ASTERISK-08654: No IP given in Console Menu if eth0 is not enabled.
ASTERISK-08655: SVN trunk is out of date
ASTERISK-08656: SVN out of date
ASTERISK-08657: Paging causes sip phone to ring instead of auto answering
ASTERISK-08658: No audio when peer sends IPV4 0.0.0.0 in sdp
ASTERISK-08659: inuse counter not decreased after hanging up a call during which caller is put on hold
ASTERISK-08660: echo while dialing through zaptel card
ASTERISK-08661: Dial's G option causes strange behavior in some circustances.
ASTERISK-08662: zap channel hangup is not detected if it occurs before the line is answered
ASTERISK-08663: Bus error when re-reading config files
ASTERISK-08664: make vim syntax highlight config files in the /etc/asterisk directory
ASTERISK-08665: [patch][moremanager branch] IAX had no manager command to list peers in proper format
ASTERISK-08666: Random segfault when two channels are simultaneously Redirect'ed into a conference
ASTERISK-08667: options from config template take precedence over per user options (iax.conf)
ASTERISK-08668: [patch] chan_cellphone - use mobile phones with Bluetooth as FXO devices in *
ASTERISK-08669: Dumps core at usecount, module format_mp3
ASTERISK-08670: EAGI buffer overflow in IPC corrupts sound transfer
ASTERISK-08671: [patch] deadlock occurs when spy leaves session using ChanSpy due to inverse locking order
ASTERISK-08672: SIP conversations drop after 30 seconds, started about 72 hours ago with no changes in network. Strangely similar to 0008193
ASTERISK-08673: [patch] IMAP Voicemail reports mailbox full if IMAP server does not report quota
ASTERISK-08674: [post-1.4][branch] CLI command audit
ASTERISK-08675: Privacy screening mode doesn't answer call before recording. Should do it.
ASTERISK-08676: Call Forward closing the second cdr
ASTERISK-08677: [patch] libedit configure for crosscompilation
ASTERISK-08678: when getting incoming call from zap you cannt put on hold or park the call
ASTERISK-08679: codec_zap.c failed to compile on asterisk 1.2
ASTERISK-08680: cant start * on PowerPC
ASTERISK-08681: New Features
ASTERISK-08682: GUI does not work properly in Opera.
ASTERISK-08683: GUI not working at all
ASTERISK-08684: Passing a channel variable to func_odbc with comma in double quotes is parsed as two values
ASTERISK-08685: DTMF modes are bot getting translated in P2P bridging mode
ASTERISK-08686: Duplication of commands created by changes in revision 47051
ASTERISK-08687: Zero length string papameters while logging in build_user function
ASTERISK-08688: pick up feature in trixbox
ASTERISK-08689: codec_zap fails to compile on Fedora Core 6
ASTERISK-08690: Incorrect SDP in header when not native bridging
ASTERISK-08691: Documentation Update for pgsql table setup
ASTERISK-08692: inUse counter not decremented after hanging up a call which is on hold
ASTERISK-08693: DTMF translation from rfc2833 to inband is not reliable (no native bridging)
ASTERISK-08694: module unload app_playback.so twice will segfault *
ASTERISK-08695: After ForkCDR AGI cant set CDR(userfield)
ASTERISK-08696: transfer is decline
ASTERISK-08697: [patch] paused status missing from realtime queue members
ASTERISK-08698: Revision 53033 displays SVN-trunk-r53002
ASTERISK-08699: gui install.html no menu , no active button
ASTERISK-08700: Asterisk accepts RTP from random endpoints
ASTERISK-08701: Pressing [Alt][F9] before the final menu pops up causes unperdicable results
ASTERISK-08702: RTP debug prints negative values
ASTERISK-08703: [patch] app_userevent sends an extra blank line (\r\n\r\n) after events
ASTERISK-08704: Selecting shutdown on AsteriskNOW Live Beta 1.4 doesn't complete
ASTERISK-08705: [patch] Channel/Thread deadlock with heavy bi-directional traffic on PRI - GLARE!
ASTERISK-08706: Asterisk doesn't see a bridged call even though the call goes through
ASTERISK-08707: glob as a file name in error message
ASTERISK-08708: [patch] asterisk -F option missing in getopt
ASTERISK-08709: Bad CALLERID after ForkCDR
ASTERISK-08710: single step to parkcall isn't working properly although regular parking and transfers does
ASTERISK-08711: [branch] need more 'relaxed' RFC2833 handling to interop with TI-based SIP-adapters
ASTERISK-08712: need more 'relaxed' DTMF INFO handling to interop with TI-based SIP-adapters
ASTERISK-08713: FIELDQTY() does not parse as expected
ASTERISK-08714: Failure to compile on Redhat Enterprise systems: ZT_TCOP_RELEASE undeclared
ASTERISK-08715: rtp.c: RTCP RR transmission error to, rtcp halted Success
ASTERISK-08716: Update Makefile to show example of 'systemname' option
ASTERISK-08717: moh-class set with SetMusicOnHold() overwrites the one set with 'm' option of Dial()
ASTERISK-08718: [patch] Asterisk to GoogleTalk client communications fail
ASTERISK-08719: [patch] User has to wait for ResponseTimeout before extension is dialed
ASTERISK-08720: Patch to correctly negotiate DTMF mode when it's set to auto or SIPDtmfMode app is used
ASTERISK-08721: can not forward Old voicemail messages
ASTERISK-08722: Created voicemail can not be forwarded to * seperated list
ASTERISK-08723: app_dial dosn't set DIALSTATUS if technology is missed.
ASTERISK-08724: prepend should not mix the current voicemail file
ASTERISK-08725: Core dumped when a Zap channel being 'Redirect'ed is hung-up
ASTERISK-08726: chan_misdn: Redirecting no. (& reason) are not set on dialout
ASTERISK-08727: ISDN user-to-user information field read/write
ASTERISK-08728: ISDN user-to-user information field read/write
ASTERISK-08729: Minor Compilation Errors on zaptel svn branch 2083
ASTERISK-08730: Zaptel is not compiling, something related to xbus-core.c - xbus-core.o
ASTERISK-08731: Errors while compiling svn 53142
ASTERISK-08732: Directed pickup fails, No target channel
ASTERISK-08733: GUI don't work on SVN version
ASTERISK-08734: core dumps on 302 forward event when forwarding ignoring set
ASTERISK-08735: [patch] AMI Status action can retunt two Responses
ASTERISK-08736: T.38 issue.
ASTERISK-08737: Add some flexibility for AgentCallbackLogin
ASTERISK-08738: Transfer and Variables
ASTERISK-08739: missing audio for sayduration parameter
ASTERISK-08740: [patch] Configurable voicemail option prompts
ASTERISK-08741: [patch] no callerid on incoming calls
ASTERISK-08742: No Idle state change message when call is transferred
ASTERISK-08743: Asterisk crashes without notice when sending a 10 digit CID to a VoIP provider (Cisco GW)
ASTERISK-08744: insecure=very not work
ASTERISK-08745: [patch] memory leaks with IAX realtime
ASTERISK-08746: clear up the description of 'sendvoicemail' and 'dialout' parameters
ASTERISK-08747: ExtensionStatus is no longer provided via the Manager interface
ASTERISK-08748: chan_skinny randomly crashing server
ASTERISK-08749: [patch] fix incorrect quotation of To: address of email notification
ASTERISK-08750: stop application
ASTERISK-08751: [patch] reloading a keyword e.g. "extconfig" prevents further reloads
ASTERISK-08752: incoming call can be routed only to one extension
ASTERISK-08753: [patch] Add Georgian support for say.c
ASTERISK-08754: problem with chan_zap or chan_iax2
ASTERISK-08755: e-mail attachment contains CRLF within MIME (base64) data
ASTERISK-08756: raw asterisk manager interface does not reply to querys
ASTERISK-08757: SpeechBackground: Stop Speech Rec after DTMF received
ASTERISK-08758: rtp.c not properly setting outbound DTMF payload type
ASTERISK-08759: DTMF not registered from called party to 3rd call party in 3-way call
ASTERISK-08760: Compilation warnings on kernel 2.4.27-3-386
ASTERISK-08761: SIP message retransmission time too short
ASTERISK-08762: configure --with-imap fails on FC6/CentOS4/RHEL4
ASTERISK-08763: [patch] Queue description text clean-up
ASTERISK-08764: Some of extensively used zaptel channels got blocked (become silent until asterisk restart)
ASTERISK-08765: Strange audio problems (distortion, truncation...) on Intel G965 / Core2Duo system
ASTERISK-08766: Calling Rules have stopped working
ASTERISK-08767: configure ignores --without-oss --without-h323
ASTERISK-08768: Garbled sound with speex and asterisk 1.4
ASTERISK-08769: ./bootstrap.sh does not run
ASTERISK-08770: Provide OPAL support for chan_h323
ASTERISK-08771: Call queues do not work when configured via web interface
ASTERISK-08772: Calling Rules do not work correctly when specified via web interface
ASTERISK-08773: chan_cellphone client
ASTERISK-08774: configure --with-snmp uses net-snmp-config --agent-libs instead of --libs
ASTERISK-08775: makeopts file missing ${prefix}
ASTERISK-08776: configure --with sqlite fails on FC6/RHEL4/CentOS4
ASTERISK-08777: Unable to use Dynamic Features
ASTERISK-08778: realtime optimisation
ASTERISK-08779: Asterisk says it cannot find a required config file (contactinfo.conf)
ASTERISK-08780: Asterisk and Zaptel have 1.4-current.tar.gz files, but not libpri
ASTERISK-08781: IRQ clashing problem
ASTERISK-08782: using AMD with no params crash asterisk
ASTERISK-08783: DUNDi ignoring incoming replies when system is busy
ASTERISK-08784: $RTPAUDIOQOS dosn't report
ASTERISK-08785: Sipura 2000 cannot authenticate when both sip identities are in use
ASTERISK-08786: ast_best_codec knows nothing about G722
ASTERISK-08787: [patch] Add confirmation of forwarded-to user via name or extension when forwarding a voicemail to another mailbox
ASTERISK-08788: install.html fails with zapscan.conf warning
ASTERISK-08789: [patch] Provide colors for daeminized asterisk
ASTERISK-08790: Realtime mode in chan_sip.c add ipsvr for table sip_conf
ASTERISK-08791: Sangoma Wanpipe Drivers unable to patch Zaptel SVN 2164
ASTERISK-08792: version.h contains ASTERISK_VERSION_NUM not prefixed with "0"
ASTERISK-08793: ExtensionState remains 'Hold' after putting a call off hold
ASTERISK-08794: [patch] Updated help text for app_record
ASTERISK-08795: [patch] poor handling of dropped IMAP connections
ASTERISK-08796: [patch] extend SMDI support to chan_sip
ASTERISK-08797: ChanIsAvail returns positive ${AVAILORIGCHAN} if checking non existent IAX peer
ASTERISK-08798: chan_sip causes spurious DNS lookups on compound hints
ASTERISK-08799: NI-2 'Operator System Access' IE (0x01) not implemented?
ASTERISK-08800: chanspy causes segfault
ASTERISK-08801: After svn update sip host not work
ASTERISK-08802: Segmentation fault in socket_process
ASTERISK-08803: Error at call from h323 to SIP
ASTERISK-08804: Error at call from h323 to SIP
ASTERISK-08805: Can't be activated without mISDN-user 1.1.0
ASTERISK-08806: Cannot make compatible if video codecs do not match and audio codecs require transcoding
ASTERISK-08807: transfered call does not edn until timeout when closed from the transferee
ASTERISK-08808: [patch] Asterisk fails to pass in-band DTMF to far end during PROGRESS state.
ASTERISK-08809: [patch] agi dumphtml has incorrect closing tag
ASTERISK-08810: Retrieve peer address of inbound call.
ASTERISK-08811: [patch] Search by first name does not match short names
ASTERISK-08812: Recordings tab does not store list of recordings
ASTERISK-08813: user pin doesnt work like expected
ASTERISK-08814: svn last check report
ASTERISK-08815: [patch] QSIG ROSE-12 and ROSE-13 support
ASTERISK-08816: [patch] Playback(<file>|noanswer) and in-band info are not working with chan_skinny
ASTERISK-08817: Going from hold to unhold status fails with Uniden phones
ASTERISK-08818: The restart text "Restarting Asterisk NOW..." might confuse someone
ASTERISK-08819: Locking strategy document
ASTERISK-08820: Most heavily used Zap channels duying (become silent) after several days of operation
ASTERISK-08821: One step parking works only in the first call
ASTERISK-08822: [patch] print the raw read/write format
ASTERISK-08823: Asterisk crashes randomly under heavy load with GSM->G.729 conversion
ASTERISK-08824: Registration returned -1: Invalid argument with Wengo
ASTERISK-08825: Bad request protocol Packet
ASTERISK-08826: Agent in queue unavailable?
ASTERISK-08827: [patch] app_queue description is missing membername parameter
ASTERISK-08828: queue show and queue show <queuename> not working
ASTERISK-08829: When caling queue, music on hold is played but not constant
ASTERISK-08830: Install is not using the right kernel version in /lib/modules
ASTERISK-08831: Support for autentication through LDAP server
ASTERISK-08832: CLI bad core show command crash Asterisk
ASTERISK-08833: Command to show zaptel verison
ASTERISK-08834: calling rules do not appear. Loading screen constantly appears in the bar above.
ASTERISK-08835: [patch] dnsmgr is not refreshed upon peer usage in realtime
ASTERISK-08836: Realtime change of host in iaxpeers does not force a change in the peers without reload
ASTERISK-08837: Calls arriving into the dialplan from chan_zap with a zero length Called Number IE fail to goto 's' priority of assigned context
ASTERISK-08838: put on field regserver, exactly the ip where sip was registered
ASTERISK-08839: chan_zap problem after reload
ASTERISK-08840: Incoming calls rejected
ASTERISK-08841: Coredump from 1.4-svn in ast_channel_free()
ASTERISK-08842: bindaddr=0.0.0.0 no bind to eth0:1
ASTERISK-08843: [patch] know which file format is currently playing
ASTERISK-08844: [patch] conf_free freed conf
ASTERISK-08845: ChanSpy Crash
ASTERISK-08846: MWI on Avaya phones does not work
ASTERISK-08847: Early media in SIP
ASTERISK-08848: Can't transfer direct to Voicemail
ASTERISK-08849: voicemail email subjetc and voice attachment id difference
ASTERISK-08850: [patch] Make sla.conf.sample easier to read
ASTERISK-08851: asterisk takes a lot of time loading/reloading if not connected to internet
ASTERISK-08852: some sound tarballs arent working
ASTERISK-08853: Source of call blown away when setting Caller ID on PRI
ASTERISK-08854: Log files are not being properly flushed
ASTERISK-08855: transfer to an IAX channel fails if transferer is P2P-bridged to a transferee
ASTERISK-08856: [patch] RTP packetization won't work under P2P bridging mode
ASTERISK-08857: [patch] French voicemail is not really french
ASTERISK-08858: [patch] followme for french
ASTERISK-08859: [patch] A few problems in safe_asterisk
ASTERISK-08860: code updates break ODBC and IMAP storage options
ASTERISK-08861: [patch] Agent logoff soft not working
ASTERISK-08862: properly include lock.h in utils.c
ASTERISK-08863: [patch] small patch for ast_translate
ASTERISK-08864: Update to dialing rule fails
ASTERISK-08865: [patch] Asterisk duplicates results for enumlookups
ASTERISK-08866: Voicemail attachments not working in asterisk 1.4
ASTERISK-08867: Compile-time failure in codec_zap.c
ASTERISK-08868: Asterisk server crash during Passive Listening - Core segmentation fault
ASTERISK-08869: Segementation fault. Crashes
ASTERISK-08870: [patch] calls in queue are blocked untill it is first call in queue
ASTERISK-08871: asterisk-gui, make checkconfig displays wrong URL
ASTERISK-08872: Zaptel TDM400P. No Analog ports detected. Duplicate lines in zapata.conf and zaptel.conf
ASTERISK-08873: core show file version empty
ASTERISK-08874: nested/redundant #if !defined(LOW_MEMORY)
ASTERISK-08875: crash svn version
ASTERISK-08876: Make fails at codec_zap
ASTERISK-08877: Not setting input[output]_device in alsa.conf causes a crash
ASTERISK-08878: Zaptel trunk doesn't compile on centos 3 (kernel 2.4.21-47.0.1.ELsmp)
ASTERISK-08879: [PATCH] Background app fails to default language if context specified
ASTERISK-08880: Placing a call on hold sends two INVITES in specific cases (with Cisco 2,811 fateway)
ASTERISK-08881: app_voicemail crashes when replication server falls down
ASTERISK-08882: Function LANGUAGE not registered
ASTERISK-08883: cannot build zaptel
ASTERISK-08884: [patch] Clearglobalvars option does not function on dialplan reload
ASTERISK-08885: asterisk sending text file attachments instead of wav attachment
ASTERISK-08886: Unable to create a custom trunk by a specified name
ASTERISK-08887: Attempting to unselect res_jabber in menuselect locks menuselect
ASTERISK-08888: coredump on incoming IAX2 call
ASTERISK-08889: [patch] adapt code to see all formats file
ASTERISK-08890: chan_skinny doesn't periodically update time on phone
ASTERISK-08891: [patch] native bridging support for chan_skinny
ASTERISK-08892: [patch] check for frame before duping it
ASTERISK-08893: Goto does not proceed to next prio if jump fails
ASTERISK-08894: Asterisk crashes as soon as chan_misdn.so is unloaded
ASTERISK-08895: Wrong FROM header
ASTERISK-08896: Voice Mail Attachments Not Displaying As Attachments
ASTERISK-08897: Core Dump in pbx_retrieve_variable
ASTERISK-08898: Phone based forward will cause Asterisk to stop responding
ASTERISK-08899: Eval leaks stack data on the end of the result string
ASTERISK-08900: AsteriskNOW IAX users are not callable
ASTERISK-08901: Zaptel fails when compiling V1.4 SVN branch
ASTERISK-08902: Extend RRMEMORY strategy to use penalty (1.4.0 and 1.2.15)
ASTERISK-08903: HDLC Bad FCS(8) on Primary D-Channel of Span 1
ASTERISK-08904: asterisk segfaulted in pbx.c
ASTERISK-08905: SIP NOTIFY messages for hints are sent with wrong request-URI
ASTERISK-08906: Segfault in res_odbc w/ pgcluster when replication server falls down
ASTERISK-08907: Segfault in cdr_odbc
ASTERISK-08908: sip CLI commands disappear
ASTERISK-08909: In use status not correct for sip queue members
ASTERISK-08910: Asterisk Crash's when callfoward from linksys 941 is activated
ASTERISK-08911: GUI access to enable debug logging
ASTERISK-08912: Segfault on transfers from an incoming IAX2 or Zap, towards a Queue with Agents, through a Local Dial
ASTERISK-08913: [patch] Background DTMF escape to given context broken
ASTERISK-08914: Call parking causes crash/deadlock
ASTERISK-08915: asterisk loosing networking
ASTERISK-08916: Asterisk crashes when a sip phone answers the Page
ASTERISK-08917: Single SIP packet can reproducibly crash Asterisk
ASTERISK-08918: [patch] Incorrect handling of zero-length frames for codecs capable of native PLC
ASTERISK-08919: AEL parses MYSQL wrong
ASTERISK-08920: Call-Limit Counter can be easily broken
ASTERISK-08921: [patch] answeronpolarityswith is not working after a reload
ASTERISK-08922: [patch] Adding language for core show channels and dumpchan
ASTERISK-08923: IF does not work when value contains : (colon)
ASTERISK-08924: make menuselect => core-sounds not ENglish => file not found
ASTERISK-08925: Channel/Thread deadlock with heavy in-bound traffic on PRI - GLARE!
ASTERISK-08926: Crash while picking up a parked call
ASTERISK-08927: [patch] update for core*
ASTERISK-08928: How to use CHANNEL(language) in extensions.conf ?
ASTERISK-08929: ooGetMsgTypeText() returns incorrect text representation on message types above OOFacility
ASTERISK-08930: Zaptel 1.2.15 not compiling on RedHat 9
ASTERISK-08931: marker bit lost in bridge_p2p_rtp_write
ASTERISK-08932: null pointer dereference in res_jabber.c after "Resource (null) of buddy ... not found"
ASTERISK-08933: GUI: hardcoded path to gui_sysinfo inside sysinfo.html
ASTERISK-08934: Crash when loading queue via realtime.
ASTERISK-08935: Not enough information about security issues.
ASTERISK-08936: Crash with SLA
ASTERISK-08937: Voicemail does not play default greeting when no options specified
ASTERISK-08938: Crash under load from multiple SIP calls
ASTERISK-08939: [patch] Zaptel build no longer honours HOSTCC
ASTERISK-08940: [patch] Memory Corruption on SMP systems causes Kernel Panic
ASTERISK-08941: race condition in sip hangup with reinvited media
ASTERISK-08942: Can't Connect two Broadvoice accounts
ASTERISK-08943: SVN 1.2 Rev 57962 Won't Build
ASTERISK-08944: 'r' option disable RTP early bridge problem
ASTERISK-08945: followme application is not respecting the channel language in order to play localized audios
ASTERISK-08946: cdr_odbc causes segfault when odbc database becomes unavailable
ASTERISK-08947: Wrong behaviour in Asterisk Base64 conversion routines
ASTERISK-08948: cant unload chan_sip
ASTERISK-08949: [patch] MYSQL Stored Procedures
ASTERISK-08950: In voicemail.conf sendmail => date english format
ASTERISK-08951: [patch] Make asterisk push an event to the manager in the case of a transfer.
ASTERISK-08952: [patch] Eyebeam cannot renew subscriptions for presence info
ASTERISK-08953: [patch] Early bridge is performed on channels with incompatible codecs when directrtpsetup=yes
ASTERISK-08954: RTPtimeout still considered in T.38 passthrough call /ast1.4.1/
ASTERISK-08955: core dump on FreeBSD 6.2 with LOW MEMORY enabled when various manager commands are issued
ASTERISK-08956: app_directory does not seem to use recorded greetings if using odbc voicemail storage
ASTERISK-08957: sip doesnt bind to all
ASTERISK-08958: vm-first spanish sound is wrong
ASTERISK-08959: app_directory crashes after recent ODBC changes
ASTERISK-08960: zaptel 1.2.15 requires kernel 2.6.17 or better for gfp_t
ASTERISK-08961: [branch] [patch] No way to create actual macros in AEL2.
ASTERISK-08962: Asterisk Registering to other SIP servers on non-default port.
ASTERISK-08963: Pattern matching occurs at every sequence number of the dial plan
ASTERISK-08964: [Patch] voicemail permissions fixes
ASTERISK-08965: vprintk availability in kernels < 2.6.9
ASTERISK-08966: compilation breaks with DBUSYDETECT_MARTIN and DBUSYDETECT_TONEONLY enabled
ASTERISK-08967: SLA continue to ring after hangup
ASTERISK-08968: Blind transfers are parsed by [default], but attended transfers work as expected
ASTERISK-08969: [patch] Asterisk leaves lingering UDP ports until no more ports available
ASTERISK-08970: [patch] Calling bad SQL statement from func_odbc causes crash
ASTERISK-08971: [branch] manager seems to not flush it's eventq like it should
ASTERISK-08972: [patch] sip attended transfer - xfersound
ASTERISK-08973: when stress-testing load/unloading modules, * deadlocks
ASTERISK-08974: [patch] added support for namealias and added support for searching by first and last
ASTERISK-08975: ZT_EVENT_REMOVED addition causes failure [patch]
ASTERISK-08976: Can't build HPEC into 1.4 branch yet
ASTERISK-08977: Jitterbuffer is abstracted from IAX2, but there is no 'core show netstats'
ASTERISK-08978: [patch] Support for Cisco 7935
ASTERISK-08979: Channel variables are not available in 'h' extension if channel goes zombie
ASTERISK-08980: soxmix removed from debian sid, need sox -m to do the same in Monitor application
ASTERISK-08981: [patch] typo in help
ASTERISK-08982: [patch] make message button work on skinny
ASTERISK-08983: Asterisk silently loses DTMF digits when sending them through ast_dtmf_stream()
ASTERISK-08984: [patch] User attributes in From tag
ASTERISK-08985: Coding problem or a bug in Asterisk(AGI)
ASTERISK-08986: pickup_exec: No target channel found for
ASTERISK-08987: [patch] OS X/gcc inline optimization incompatability in 1.2.16
ASTERISK-08988: [patch] insecure && ~sipregs == Failed to authenticate
ASTERISK-08989: [patch] fix totalAnalysisTime to handle periods of no channel activity
ASTERISK-08990: [patch] Eliminate the comment related global vars from main/config.c
ASTERISK-08991: Snom Call Pickup Of Subscribed Extensions
ASTERISK-08992: make the app_dial resources configurable when returning back from a parking timeout
ASTERISK-08993: [Patch] SMDI features
ASTERISK-08994: Asterisk crash related to the manager and call origination
ASTERISK-08995: Voice mail prompts in languages other than English play English digits and dates
ASTERISK-08996: [patch] COMPLETECALLER event logs channel identifier instead of membername
ASTERISK-08997: Misleading message for dealing with temporary greetings
ASTERISK-08998: Can't repark calls using one touch park
ASTERISK-08999: Greetings stored in ODBC voicemail don't play [patch]